On Fri, 7 May 2004, Brian Cuthie wrote: > It seems that each time I get a new checkout of * from CVS my Cisco 7960 > works worse than before. I know this stuff's in flux, so I mention this > in case it's news. Anyone else having trouble? What I'm seeing (er, > hearing) is really choppy audio. The previous version I had installed > had fairly frequent audio dropouts (not present when I make the same > calls through the same * box using a TDM400P interface).
I had jittery audio with dropouts on a 7960 with SCCP, and started testing SIP hoping it would be better (based on the reports of the SIP-to-IAX2 timestamping issue). Here's my experience: * As Brian mentions, when the other end of the call is from a non-VoIP path (e.g. Zaptel interface) the audio is fine. * Calls over IAX start out okay, but within a few seconds the audio starts jittering. It gets progressively worse until about a minute into the call (often less), by which time audio is unintelligible. Calling the same number over the same IAX connection from an analogue phone attached to a SIP-image ATA-186 which in turn is plugged into the "PC" port of the same 7960 gives perfect audio. * Calls over SIP are stable; I had an intermittent problem where audio into the 7960 would stop completely for up to three seconds, but that seems to be gone after doing a CVS update. Side note: when I had this audio dropout problem, making the same call without * in the audio path (by using canreinvite=yes and removing t and T from Dial) resulted in perfect audio. I'll try someone's suggestion to disable the jitterbuffer to fix the IAX2 problem, but I thought that the jitterbuffer was supposed to help this kind of problem... Besides, the same call over an ATA or using X-lite is perfect. Before anyone jumps in, yes, as soon as I can get there I will hit the bug tracker. Cheers, Vic Cross PS: I know that folks generally dislike 'me too' messages, but this time Too Bad -- I'm trying to provide more info to help anyone that might be working on problems. <rant> I hope that Iain was exaggerating when he described his bug-reporting experience. Many * users are unable to commit the time to poring over hundreds of lines of uncommented C code and ethereal traces with thousands of packets captured. So, as our way of trying to help, we provide e-mails like this either in response to or as an attempt to gather more information about the problem. To try and get people talking about a problem. How is does it help to jump on someone who is trying to get resolution to a problem -- by driving them toward OpenPBX or VOCAL? A few former colleagues of mine may soon be about to learn (unfortunately) that you can only piss off a customer so many times. To the Asterisk developers, bug marshals, and coders: I am jealous of you! You've created a wonderful thing. I'd love to be able to spend the amount of time I'd like to on Asterisk. I'd love to be able to do more to fix bugs and develop features. But I can't. Don't think less of me because of that. </rant> VC _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
